Fixed IP networks were originally designed to carry “best effort” traffic where the network makes a “best attempt” to deliver a user packet, but does not guarantee that a user packet will arrive at the destination. IP networks need to support various types of applications. Some of these applications have Quality of Service (QoS) requirements other than “best effort” service. Examples of such applications include various real time applications (IP telephony/voice, video conferencing), streaming services (audio or video), or high quality data services (browsing with bounded download delays).
Although fixed, wireline IP networks are well-established, new and different challenges face IP communications in mobile, wireless communication networks. Consider for example an IP telephony session between a mobile User-A and a User-B, where User-A accesses an IP backbone through a local, radio access, mobile communications network. Examples include a Global System for Mobile communications (GSM) or a Universal Mobile Telecommunications System (UMTS) network.
Quality of service for an IP telephony session can be characterized in terms of bandwidth, packet loss, delay, and jitter. As far as the users are concerned, the perceived quality of service will likely depend on the service provided by the radio access network e.g., a UMTS or GSM/GPRS network. The radio interface is the most challenging interface in the communication in terms of providing a certain capacity and coverage at a particular quality of service. Radio bandwidth is limited, and therefore, the capacity of the system to deliver services to a certain number of users is necessarily limited. And it typically is not an option to achieve a particular quality of service for one or a few users simply by allocating more radio bandwidth. Coverage in a radio mobile radio system is the maximum distance over which a radio base station can reliably communicate with a mobile radio. That coverage is limited by maximum signal transmission levels, fading, path loss, interference, and other impediments to radio transmission.
Third generation mobile communications systems, like a Universal Mobile Telephone communications System (UMTS), provide mobile radios with the ability to conduct multimedia sessions where a communication session between users may include different types of media. A very important medium to support in multimedia sessions is voice.
So there is a need to provide resource-efficient, packet-based conversational (e.g., voice) multimedia services that can be delivered at a quality of service and cost comparable to a traditional circuit-based voice service. Even though the idea of a conversational IP multimedia system (IMS) service is desirable, a practical implementation of a conversational IMS service requires overcoming several technical hurdles before the idea becomes a commercial reality. Conversational IMS services should deliver high speech quality both in terms of fidelity and low delay. Connection set up and service interaction times should be reasonably fast. Packet-based speech must meet very strict delay requirements. For example, a new speech packet in a UMTS may need to be delivered every 20 milliseconds. The radio spectrum must be used efficiently. Services must cover a wide geographic area and be able to service roaming mobile users. Although wire-line access networks permit over-provisioning, wireless networks cannot afford that luxury because of limited radio bandwidth and the need to support user mobility.
One obstacle to efficient communication from a capacity and coverage point of view is the size of the speech packet header relative to the speech packet payload, i.e., the speech content. As shown in FIG. 1, speech packets typically have a long header and a relatively short speech payload. Speech samples are typically sent very frequently, e.g., every 20 ms, so the payload is small. But the header information is not reduced just because the payload is small. An example speech packet header suitable for use in a UMTS contains several packet handling protocol headers—IP header information, User Data Protocol (UDP) header information, and Real Time Protocol (RTP) header information. Each protocol contributes to a longer speech packet header.
Even though the speech packet payload is small, the real time delay requirements for packet-based speech service, (e.g., one speech packet every 20 ms), means there is no time to consolidate multiple payloads into one packet. Such consolidation might be possible for a voice streaming application having less stringent, real time delay requirements. Transmitting a large amount of header information for a relatively small amount of speech content results in low throughput. Too much of the radio bandwidth is needed to transmit “overhead” header information. Using that limited bandwidth to transmit the overhead header information decreases the capacity of the mobile communications system to transmit payload information over the radio interface. Because the mobile terminals have a finite transmit power, using power to transmit overhead information means that less power is available to transmit payload information. Lower power per bit means the physical distance over which a radio communication can take place is reduced, i.e., reduced coverage. Accordingly, the amount of header overhead must be reduced to increase capacity and/or coverage.
Header compression may be used to reduce the amount of header information that must be transmitted with each speech packet over the radio interface. Many header compression techniques generally work by sending only new or changed header information sometimes called “delta” information. The delta information is much less than the normal header information, so much less radio bandwidth is needed to transmit the delta (compressed) header as compared to the full (uncompressed) header.
But there are times when more information must be sent. One such time is when a speech connection is being established. At that point, the full (uncompressed) header must be sent in order to establish the foundation for sending just the delta (compressed) header. Even the amount of delta information varies during the communication. Normally, the delta information is small. But on occasion, it may temporarily increase in size during the communication. If the radio resources for a speech connection are established based on the full header, those resources will later be wasted when the header is sent in compressed form, e.g., just the delta is sent. Alternatively, the radio resources may be established based on the compressed header. But then there is no provision for handling the additional data presented by an uncompressed header or a partially compressed header.
The present invention increases the capacity and coverage of a packet-based conversational service in a mobile communications system by efficiently provisioning resources for speech packets that have dynamic or different length headers during the lifetime of a connection over which the service is provided. A radio bearer, which can be viewed as a logical connection, is established over the radio interface between a mobile radio and a radio network to support a packet-based conversational service. The radio bearer is configured with radio resources to deliver a particular quality of service assuming that the headers will be compressed. The length of or amount information in each packet to be transmitted via the radio connection is detected. Those packets which are longer or have more information because there is no header compression or lesser header compression are detected. Processing is performed to minimize or reduce the impact on system capacity or coverage when transmitted over the radio bearer.
In one example embodiment, reduced impact is achieved by segmenting and buffering longer packets so that the data rate is kept constant or below a maximum rate over the radio interface. The segments are transmitted at the same rate as compressed packets. A buffer management scheme is advantageously employed to manage the segments and packets in the buffer. If too many packets accumulate, old packets are discarded. Bandwidth requirements are maintained, and the administrative cost associated with buffering and buffer management is relatively low.
In another example embodiment, reduced impact is achieved by reconfiguring the radio bearer or connection used to transmit the increased or decreased amount of header data for the different length packets at substantially the same data rate. In this way, strict delay requirements are met, and no packets are discarded. The cost associated with this example embodiment includes increasing the radio bearer bandwidth for longer packets, temporarily diminishing the capacity of the system. There is also an administrative cost and delay associated with reconfiguring the radio bearer connection.
Both example embodiments save system capacity as compared to the situation where radio resources are allocated to transmit full, uncompressed packet headers. The first example embodiment also maintains system coverage but at the cost of some discarded packets buffered for too long. The second example embodiment does not preserve system coverage as well as the first example embodiment, and therefore, may be less preferred.
In both example embodiments, the information can be sent in a transparent mode or in a non-transparent mode. In the transparent mode, there are no explicit information that informs the receiving entity where a speech packet starts or stops. Instead, this packet start/stop information is provided implicitly by fitting the packets exactly into one of plural known formats used by the transporting radio bearer. The transparent mode has a higher capacity than the non-transparent mode, but requires a limited number of known sizes of packet information. The non-transparent mode, on the other hand, lacks this requirement but suffers from somewhat lower capacity.
In another aspect of the invention, a packet protocol may further be employed between the mobile radio and the radio network that does not require transmission of a checksum on the whole packet, thereby providing unequal error protection. Unequal error protection has been used for circuit-switched voice services to improve spectrum efficiency and similar advantages could also be achieved for a packet switched conversational service. Such unequal error protection may be implemented in Internet Protocol version 6 (IPv6) communications using a modified version of User Datagram Protocol (UDP) commonly called UDP lite.